Freddi Hansen
2012-Mar-18 18:53 UTC
[asterisk-users] INVITE retransmission by 1.8... (Steve Murphy)
> I have a site that moved to the latest 1.8 revision, and began to > have problems with phones in "far away places" (South America, > and the MidEast). > > What I see is that when a Dial() is issued, the sip channel driver > sends out an INVITE to the phone. Very soon thereafter, Asterisk > retransmits the INVITE. The phone sends back a 100 Trying, and > then, usually, a 400 response. I may be misinterpreting what I see, > but I *think* that the response from the phone is delayed just enough > to invoke the retransmission. The phone responds to the second invite > with a 400 code, which Asterisk interprets as an error, and the call > immediately > goes into hangup. > > Has anyone else seen this? It doesn't happen all the time, and only > with certain > phones. It comes and goes, but when it comes, phones become > unreachable. It > seems to be in this state the majority of the time for certain phones. > While most > phones seem far away, some are in Florida. > > We replaced the 1.8 version of Asterisk with a 1.6.2 version, and the > issue has > gone away. I know, I know, I could give more detail, fill out a bug > report, but > this is the early stages. I may be misinterpreting what I'm seeing. > > Anyone else seen this sort of thing? Any words of wisdom?hi, one of our gateways is used for SIP over satelite links and we se the same thing on default installation. The fix is to change chan_sip.c #define DEFAULT_RETRANS to a higher value, we use 3000. The retransmit timer at the far end (pap2t) is increased to 3 times its standard values. It probably breaks some sip specs but its needed to keep it working when roundtrip gets to big. Freddi
Stefan Schmidt
2012-Mar-19 08:55 UTC
[asterisk-users] INVITE retransmission by 1.8... (Steve Murphy)
Am 18.03.12 19:53, schrieb Freddi Hansen:>> I have a site that moved to the latest 1.8 revision, and began to >> have problems with phones in "far away places" (South America, >> and the MidEast). >> >> What I see is that when a Dial() is issued, the sip channel driver >> sends out an INVITE to the phone. Very soon thereafter, Asterisk >> retransmits the INVITE. The phone sends back a 100 Trying, and >> then, usually, a 400 response. I may be misinterpreting what I see, >> but I *think* that the response from the phone is delayed just enough >> to invoke the retransmission. The phone responds to the second invite >> with a 400 code, which Asterisk interprets as an error, and the call >> immediately >> goes into hangup. >> >> Has anyone else seen this? It doesn't happen all the time, and only >> with certain >> phones. It comes and goes, but when it comes, phones become >> unreachable. It >> seems to be in this state the majority of the time for certain phones. >> While most >> phones seem far away, some are in Florida. >> >> We replaced the 1.8 version of Asterisk with a 1.6.2 version, and the >> issue has >> gone away. I know, I know, I could give more detail, fill out a bug >> report, but >> this is the early stages. I may be misinterpreting what I'm seeing. >> >> Anyone else seen this sort of thing? Any words of wisdom? >There is a option in sip.conf you should use for this and its called T1 the normal retransmit timeout is either T1*2 counting up for each retransmission (T1*4,*8,...T1*64) or it depends on the maxms values which a qualify gives you. So if you enable qualify and you have values like 300 ms to the phone the retransmission should honor this, or as i said just change the t1 value from 500 ms (default) to 1000 or 2000 ms for example. then asterisk should not try to retransmit the next invite so early. best regards stefan> hi, > one of our gateways is used for SIP over satelite links and we se the > same thing on default installation. > The fix is to change chan_sip.c > #define DEFAULT_RETRANS to a higher value, we use 3000. > > The retransmit timer at the far end (pap2t) is increased to 3 times its > standard values. > > It probably breaks some sip specs but its needed to keep it working when > roundtrip gets to big. > > Freddi > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Teamleiter VOIP // voip at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH - a Tele2 Company // Donau-City-Strasse 11 // A-1220 Wien // -------------------------------------------------